Answering to your questions: Why are we using circular convolution in DSP? In DSP we normally deal with finite length discrete sequences (even if the signal under study is infinite we can only ...

Clements and Pease have shown that causal infinite-duration impulse responses can also have Fourier transforms with generalized linear phase. The corresponding system functions, however, are not ...

The kind of filter you are looking for is a notch filter. Using filter design toolbox in Matlab you can get it as I'm showing you in the following picture: This has to be done in z-plane so there ...

Welcome to DSP! First of all I would recommend reading chapter 2, specially 2.2 to 2.5 of Discrete Time Signal Processing 3rd edition by Oppenheim and Schafer. Then 3.2 and 5.2 of that very same book. ...

Every symbol in the channel will be affected by ISI from previous symbols and next symbols. To understand this you must notice that there's a propagation delay from transmitter(TX) to receiver(RX). ...

Since nobody has answered my question yet. I will try to partially answer it myself. However, even though I accept it as the final answer I can revoke this if another user elaborates a much better ...

Even though @Ben has already given a wonderful answer I would like to add that there's a technique called zero-phase filtering which allows you to have no phase delay at all. However, as I don't ...

Normally the higher the carrier frequency the more it will be absorbed by the environment (for instance air, water) and the smaller the propagation distance. I think this happens because the smaller ...

It seems that you are looking for books to learn about multiple topics. Skimming through your question I think these are the topics: Aspects of signals (signal mean, its power,amplitude,frequency,etc ...

If I were you I would define my sampling period as the period present in the interval with the highest sampling frequency. Then I would try to interpolate the sections of the signal that were ...

You must consider the microphone sensitivity while calibrating the recording system if you want to have an idea of what your digital samples actually meant because once they are digitized this ...

So far I haven't found a way to compute the circular convolution (of size N for sequences greater than N) that doesn't involve applying linear convolution + aliasing. Which is my original question ...

You could use correlation between the two signals or a matched filter in which you set the filter coefficients as the samples from the selected segment inverted in time. In matlab it would be ...

The success of your camera depends on too many factors! and they're out of the scope of this page. If the dynamic range requirements of your application are not too demanding then you could take ...

Yes you should definitely use FSK but it has to asynchronical and here is why. Demodulating FSK synchronically is called coherent demodulation. Demodulating FSK coherently requires incoming carrier ...

If you studied the materials I recommended you on comments to your previous question. You would notice that FFT algorithm takes advantage of certain symmetries to make computations faster. If you don'...

Assuming we are working with an ideal cosine that has a frequency response that is a perfect delta, then if the original signal is at $18 kHz$ and it is sampled at $27kHz$ then there will be aliasing (...

This is the place where I learned what you are asking: MIT Opencourseware 6.008 Digital Signal Processing . See lecture 18. And if you have the chance also see lectures 19 and 20. You are trying to ...